snd_interval_list() expected a sorted list but did not document this, so
there are drivers that give it an unsorted list. To fix this, change
the algorithm to work with any list.
This fixes the "Slave PCM not usable" error with USB devices that have
multiple alternate settings with sample rates in decreasing order, such
as the Philips Askey VC010 WebCam.
http://bugzilla.kernel.org/show_bug.cgi?id=14028
Reported-and-tested-by: Andrzej <adkadk@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a logic error in the range check of the input level control that
would prevent setting any volume less than the maximum.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modify loops in such way that the register value is checked also after
the timeout condition, just in case the heavy interrupt load etc. caused
the thread to sleep for the time period exceeding the timeout value.
While at it remove an extra ALI_STIMER read from snd_ali_stimer_ready().
Reported-by: Jack Byer <ojbyer@usa.net>
Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are many variants of Toshiba laptops with ALC268 codec, and
it seems that a few of them don't work with model=toshiba preset
since they have the secondary ALC268 codec just for HDMI output.
This is a regression due to the previous clean-up work to merge all
Toshiba quirk entries into a single check.
This patch adds the identification of such laptops to apply the
standard BIOS-probing method. Unfortunately, Toshiba laptops have
all the same PCI SSID, so we need to check the codec SSID to identify
each device.
Tested-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The BIOS pin configs are in fact correct and shall not be overwritten.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP Compaq 6530s and 6531s internal speaker is silence or becomes silence
within 1 minute after fresh boot. It is found that pin 0x1c must be set to
PIN_OUT mode to make the speaker work. This is weird - line-in pin 0x1c and
speaker pin 0x16 seem to be unrelated.
The codec differences before/after patch are:
@@ Node 0x17 [Pin Complex] wcaps 0x40020b:
Pin Default 0x41a6e130: [N/A] Mic at Ext Rear
Conn = Digital, Color = White
DefAssociation = 0x3, Sequence = 0x0
Misc = NO_PRESENCE
- Pin-ctls: 0x24: IN
+ Pin-ctls: 0x40: OUT
@@ Node 0x1c [Pin Complex] wcaps 0x40018d:
Pin Default 0x41813021: [N/A] Line In at Ext Rear
Conn = 1/8, Color = Blue
DefAssociation = 0x2, Sequence = 0x1
- Pin-ctls: 0x24: IN VREF_80
+ Pin-ctls: 0x40: OUT VREF_HIZ
Unsolicited: tag=00, enabled=0
Connection: 1
0x24
Tests show that it won't impact (external) Mic recording.
Reported-by: "Lin, Ming M" <ming.m.lin@intel.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269 and ALC861-VD parsers override the ADC definitions
unconditionally without checking the spec definition. This causes
the problem when any inconsistent ADC is set up in the device quirk
(like ALC272 with digital-mic).
This patch avoids the overriding by adding the proper checks.
Reference: Novell bnc#529467
https://bugzilla.novell.com/show_bug.cgi?id=529467
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without the initialization of vmaster NID, the dB information got
confused for ALC269 codec.
Reference: Novell bnc#527361
https://bugzilla.novell.com/show_bug.cgi?id=527361
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Read buffer overflow
ALSA: hda: Correct EAPD for Dell Inspiron 1525
ALSA: hda: warn on spurious response
ALSA: hda: remember last command for each codec
ALSA: hda: read CORBWP inside reg_lock
ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_io
ALSA: hda: take cmd_mutex in probe_codec()
ALSA: hda: track CIRB/CORB command/response states for each codec
ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527
Check whether index is within bounds before testing the element.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 24918b61b5 statically changes
the model from dell-bios to dell-3stack to solve the sound decreasing
regression (http://lkml.org/lkml/2008/9/12/203), however it leads to another
problem that the 2nd headphone jack doesn't work
(https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3987). So I think
the commit 249**2dc is just a workaround. I would like to give a true solution
here.
The datasheet for STAC9228 says, GPIO2 is the same pin as VOL DOWN, and
the EAPD pin is GPIO0. This is why the sound decreases if we set EAPD as
GPIO2. This patch changes EAPD to GPIO0 to solve the problem.
Signed-off-by: Chengu Wang <wangchengu@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This converts the last CORBWP access outside of reg_lock.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just for safety. azx_init_cmd_io() and azx_free_cmd_io() may be
called when switching to single command mode.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that each codec will have its own module, it is possible
for the user to load one codec while another one is running.
So cmd_mutex would be a safe addition to probe_codec().
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently we hit a bug in our dev board, whose HDMI codec#3 may emit
redundant/spurious responses, which were then taken as responses to
command for another onboard Realtek codec#2, and mess up both codecs.
Extend the azx_rb.cmds and azx_rb.res to array and track each codec's
commands/responses separately. This helps keep good codec safe from
broken ones.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use model=lenovo instead of model=dallas for Toshiba Satellite A135-S4527
with ALC861-VD codec.
Reference: Novell bnc#526325
https://bugzilla.novell.com/show_bug.cgi?id=526325
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check that the result of kzalloc is not NULL before a dereference.
The semantic match that finds this problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@@
expression *x;
identifier f;
constant char *C;
@@
x = \(kmalloc\|kcalloc\|kzalloc\)(...);
... when != x == NULL
when != x != NULL
when != (x || ...)
(
kfree(x)
|
f(...,C,...,x,...)
|
*f(...,x,...)
|
*x->f
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The name buf with size 16 is too short for some codec names, e.g.
truncated like "ALC861-VD Analo". Now the size is doubled.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
mpu_synth_info[m].name is a char[30], and the minimum length of the data
written by sprintf is 31 bytes including terminating null.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DSPVersion is declared as char[3], but the sprintf writes at least 4 bytes
including terminating null.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
e->sad[] is declared with size ELD_MAX_SAD=16, but the guard
allows range 0-31.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* fix/hda:
ALSA: hda - Fix mute control with some ALC262 models
ALSA: hda - Restore GPIO1 properly at resume with AD1984A
ALSA: hda - Use snprintf() to be safer
The master mute switch is wrongly implemented as checking the pointer
instead of its value, thus it can be never muted. This patch fixes
the issue.
Reference: Novell bnc#404873
https://bugzilla.novell.com/show_bug.cgi?id=404873
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
This adds support for Native Instrument's freshly announced Audio2DJ
sound device hardware. Version number bumped to 1.3.19.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fix 79452f0a28 introduced another
bug due to the missing offset for the overlapped hwptr.
When the hwptr goes back to zero, the delta value has to be corrected
with the buffer size. Otherwise this causes looping sounds.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the logging functionality to xrun_debug to record the hwptr
updates via snd_pcm_update_hw_ptr() and snd_pcm_update_hwptr_interrupt(),
corresponding to 16 and 8, respectively.
For example,
# echo 9 > /proc/asound/card0/pcm0p/xrun_debug
will record the position and other parameters at each period interrupt
together with the normal XRUN debugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PLL was not being enabled when it was not bypassed. This patch
enables the PLL when it is used. Additionally, it disables the PLL
when it is bypassed.
Without this patch, the audio on TI DM646x EVM and DM355 EVM
does not work properly. The bit clocks and the frame sync signals
from the codec are not correct and hence the playback/record are faster
than usual for most sample rates. The reason for this was that the PLL
was not enabled when it was not bypassed.
Tested on DM6467 EVM, playback tested on DM355 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit 099db17e66 introduced a
regression at suspend/resume where the GPIO1 bit isn't properly
restored, thus the speaker output gets muted initially after resume.
The fix is simple, use the cached write for storing GPIO data.
Reference: Novell bnc#522764
https://bugzilla.novell.com/show_bug.cgi?id=522764
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a few uninitialized error checks that were introduced recently
mistakenlly during the clean-up:
sound/pci/ctxfi/ctamixer.c: In function ‘get_amixer_rsc’:
sound/pci/ctxfi/ctamixer.c:261: warning: ‘err’ may be used uninitialized in this function
sound/pci/ctxfi/ctamixer.c: In function ‘get_sum_rsc’:
sound/pci/ctxfi/ctamixer.c:415: warning: ‘err’ may be used uninitialized in this function
sound/pci/ctxfi/ctsrc.c: In function ‘get_srcimp_rsc’:
sound/pci/ctxfi/ctsrc.c:742: warning: ‘err’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- E3500 report cval->max more than it actually can handel, so if you
set 95% capture level it will be silently muted.
- Betwen cval->min and cval-max(real) is 2940 control units,
but real are only 7 with cval->res = 384.
- Alsa can't handel less than 10 controls, so make it more
and set cval->res = 192.
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VMware tends to report PCM positions and period updates at utterly
wrong timing. This screws up the recent PCM core code that tries
to correct the position based on the irq timing.
Now, when a backward irq position is detected, skip the update
instead of rebasing. (This is almost the old behavior before
2.6.30.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* fix/misc:
ALSA: ca0106 - Fix the max capture buffer size
ALSA: OSS sequencer should be initialized after snd_seq_system_client_init
ALSA: sound/isa: convert nested spin_lock_irqsave to spin_lock
Linus Torvalds